Digital FIR Filter using Frequency Sampling Method
This is the second method of FIR Filter design. The steps of finding the frequency response of the input is same as that of the windowing method. Then, DFT of the frequency response is calculated. The values are adjusted depending on the type of filter i.e. low pass filter or high pass filter. Inverse of the frequency sampled signal gives the desired filter design. The designing and plotting of magnitude response was done on Scilab.
in FSM calculations are reduced
ReplyDeleteWhen the desired frequency response characterization of the FIR filter is narrowband, most of the coefficients H[k] are zero. The corresponding filter sections can be eliminated and only the filters with non zero coefficients need to be retained. The net result is a filter that requires fewer computations
DeleteWhat was the sampling frequency used?
ReplyDelete8000 Hz
DeleteFSM is easier to implement
ReplyDeleteYes because it requires less computations
DeleteThis comment has been removed by the author.
ReplyDeleteGood post
ReplyDeleteThank you
DeleteSimplest and direct technique for fir filter design
ReplyDeleteYes
DeleteRequires N multiplications per output
ReplyDeletesample
Frequency sampling realization is computationally more efficient than direct form realization
ReplyDeleteFrequency sampling realization is realization of FIR filter using combination of FIR and IIR filters
DeleteIt has time aliased response, if undersampled
ReplyDeleteWhen a signal is undersampled in the frequency domain, its spectrum has overlapping tails. That means that it is no longer possible to correctly reconstruct the time-domain signal.
DeleteThe samples obtained are identified as DFT coefficients.
ReplyDelete